Responsibilities :

- Implement core voice capabilities using FreeSWITCH, Kamailio/OpenSIPs, and RTPEngine.

- Build and optimise SIP call routing logic, RTP media relays, failover mechanisms, and NAT traversal.

- Develop and manage configurations for scalability, codec negotiation, and SIP trunk registration.

- Implement and test features like call recording, IVR, voicemail, and DTMF detection.

- Monitor live traffic and participate in 24x7 on-call rotation for critical escalations.

- Collaborate with QA on stress/load testing and with Backend teams on provisioning APIs.

- Document design decisions, configurations, and troubleshooting runbooks.

Requirements :

- Strong systems programming and debugging skills in C/C++.

- Good scripting/debugging skills (Bash, Python, or Lua for FreeSWITCH modules).

- Proficiency with diagnostic tools (Wireshark, tcpdump, etc).

- Experience working with geographically distributed infrastructure or HA deployments.